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October 28 Linksys and Sipura adapter users - check your RTP Packet Size and Network Jitter LevelLinksys and Sipura adapter users - check your RTP Packet Size and Network Jitter LevelNovember 16, 2007 at 2:04 pm · Filed under Asterisk, Elastix, FAX, FreePBX, Linksys, PBX in a Flash, Sipura, Telephone, Trixbox, VoIP ·Tagged Asterisk, Elastix, FAX, FreePBX, Linksys, PBX in a Flash, Sipura, Trixbox, VoIP Edit: Reader Christopher Woods notes in a comment that the following is also applicable to at least some models of Linksys phones, e.g. SPA942 and SPA962. Do you use a Linksys or Sipura VoIP adapter? Do the people you are talking to ever complain about your voice breaking up, or missing or dropped syllables, or unexplained clicks or noise? There is an obscure setting in Linksys/Sipura VoIP adapters that is usually set incorrectly for most applications, at least on a factory-fresh adapter. Go to the SIP tab and check the RTP Packet Size - for most users, it should be set to 0.020 rather than the factory preset of 0.030. If you are running a connection where latency is critical (say you have a cable or satellite box that requires a phone connection to “phone home”, or you are trying to use a FAX machine) then you may even wish to set this to 0.010, which further reduces latency, at the expense of using a bit more bandwidth. In any case, the default 0.030 is not the correct setting when using the most commonly-used codecs. For more discussion of this issue, see this thread at DSLreports.com, which discusses how the RTP Packet Size and Network Jitter Level settings can be tweaked to achieve lower latency, along with the tradeoffs. Be aware that the RTP Packet Size setting is found under the SIP tab, and that setting is applied to all lines served through that adapter. However, the Network Jitter Level can be set individually for each line, under the Line tabs. One interesting comment in the above-mentioned thread is that if a provider forces you to use a low-bandwidth codec, decreasing the RTP Packet Size may increase the quality of your calls, but again at the expense of increasing bandwidth used. Changing the RTP Packet Size on one VoIP adapter resolved a few strange issues with audio quality. In this case the adapter was being used to connect to an Asterisk box on the same local network, so bandwidth usage wasn’t an issue. We set the RTP Packet Size to 0.020 and the Network Jitter Level to low, and it made a noticeable difference in the reduction of strange noises and breakups heard by the party on the other end of the conversation. However, changing the Network Jitter Level isn’t as critical as changing the RTP Packet Size, and in fact, changing the Network Jitter Level may be entirely the wrong thing to do on certain types of connections (probably not a good idea if your adapter is connected through a Wireless ISP, for example). I must thank Paul Timmins for being the first to point out that the Linksys PAP2 has a default packet size of 0.030, which is incompatible with the uLaw (G711u) codec (or at least in violation of the standard). With that lead, I then discovered other articles (including the discussion thread linked above) that said essentially the same thing. So check those adapter settings, folks! (And by the way, this advice probably does apply to some other makes of VoIP adapters, and even some IP telephones, but since I don’t have any readily available to look at, I can’t say for sure. If you know of any others that need to have a similar setting tweaked, please feel free to add a comment to this post). August 06 Weather CAMS AzoresCLICAR AKI PRA VER
June 13 Unlimited Incoming/Outgoing cellphone calls with Asterisk
Unlimited Incoming/Outgoing cellphone calls with Asterisk Quick note! This guide is incomplete. If you have a decent understanding of linux and asterisk, you can probably fill in the huge gaping holes. Otherwise, I'd stay far far away! At the very least, don't try this on your production box! You will need a cellphone with bluetooth. If it does not have bluetooth, it will not work, simple as that. Also not all phones will work. Personally, I'm using a Motorola Razr V3 (also tested with a Motorola V551). You also need a usb bluetooth dongle with a CSR chipset. Broadcom will not work. I would buy this in a regular retail store so it can be taken back if it has the wrong chipset. I researched this, found the Linksys USBBT100 which has the correct chipset and great range, and ordered it. When I got it, it was a rev2 with a different chipset. The one I'm using now is a Belkin F8T001V ordered from Circuit City here. Install Asterisk@Home 2.2 (instructions should be similar, if not identical, for 2.5). login: root password: password Change the following passwords: passwd passwd admin passwd-maint passwd-amp passwd-meetme Setup Networking yum -y update cd /usr/src/zaptel make install-udev yum install bluez-libs-devel Is this ok [y/N]: Y reboot rebuild_zaptel reboot genzaptelconf cd / wget http://www.thetechguide.com/howto/asterisk/bluetoothfiles.tar.gz tar xzf bluetoothfiles.tar.gz Ignore errors about time stamp in future cd /usr/src/asterisk make clean make make install If you see a message displaying WARNING WARNING WARNING pertaining to /usr/lib/asterisk/modules, you can ignore it, those modules are supposed to be there. chkconfig bluetooth on service bluetooth start Now run hcitool scan. You will see output similar to this: [root@asterisk1 asterisk]# hcitool scan Scanning ... 00:12:8A:C7:DA:7C Motorola Phone The 00:12:8A:C7:DA:7C is the MAC address of the phone. Now, we need to edit your bluetooth.conf file to use this address: cd /etc/asterisk nano bluetooth.conf Go to the bottom, you will see this: ;; A Motorola V551 [00:12:8A:C7:DA:7C] name = Motorola type = AG channel = 3 autoconnect = yes Replace [00:12:8A:C7:DA:7C] with the MAC address you got from hcitool scan (keep the [] brackets surrounding it). Now, another thing that may cause you problems, is the "channel = 3" part. In the bluetooth.conf, it says to run sdptool search --bdaddr xx:xx:xx:xx:xx:xx 0x111F to find the correct channel number. For mine it reported channel 7, but would not work for me unless set to 3. In case this part is important for you, here's how to see what channel it thinks you should use (I'd try it with channel = 3 first, if that doesn't work then try this, if that doesn't work then start at 1 and work your way up to 13 and see if you get anything). Once you're done editing the file, hit Ctrl-X to exit, Y to save, and the enter key to keep the same filename (so Ctrl-X, Y, enter). The xx:xx:xx:xx:xx:xx is the MAC address (the same one you got from hcitool scan), so in our case the command will be sdptool search --bdaddr 00:12:8A:C7:DA:7C 0x111F. Your phone does NOT have to be in Find Me mode to run this tool. Just make sure it's on and the bluetooth service is running. In my case, the output is: [root@asterisk1 asterisk]# sdptool search --bdaddr 00:12:8A:C7:DA:7C 0x111F Class 0x111F Searching for 0x111F on 00:12:8A:C7:DA:7C ... Service Name: Hands-Free voice gateway Service Description: Hands-Free voice gateway Service Provider: Cingular Service RecHandle: 0x10007 Service Class ID List: "" (0x111f) "Generic Audio" (0x1203) Protocol Descriptor List: "L2CAP" (0x0100) "RFCOMM" (0x0003) Channel: 7 Language Base Attr List: code_ISO639: 0x656e encoding: 0x6a base_offset: 0x100 code_ISO639: 0x6672 encoding: 0x6a base_offset: 0xd800 code_ISO639: 0x6573 encoding: 0x6a base_offset: 0xd803 code_ISO639: 0x7074 encoding: 0x6a base_offset: 0xd806 Profile Descriptor List: "" (0x111e) Version: 0x0101 The part that interests us is the line right under ""RFCOMM" (0x0003)" which says "Channel: 7". So in /etc/asterisk/bluetooth.conf, you want to change "channel = 3" to "channel = 7" (again, note that this did NOT work for me, I had to have it on channel 3). At this point I rebooted, to make sure everything would start up automatically: reboot Once it comes back online, Asterisk will start up and attempt to connect to your phone. The first time this happens, it will ask if you want to connect (say yes) and ask for a PIN. The PIN is 1234. My phone kept asking if I wanted to connect each time asterisk was rebooted or the phone was powered on, you should be able to go in your phone menus and set it to just automatically connect and it won't bother you again. Next up, actually making and receiving calls and troubleshooting. Click here for the second page, also still in progress (i.e., incomplete). April 15 procurar LOGINSBugmenot.com - login with these free web passwords to bypass compulsory registrationFind and share logins for websites that force you to register:Jimmy BLOG |
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